similar to: Setting Appearance on Outbound Calls?

Hi All - I've got a load of Polycom phones, and for the most part, I think they're great, but one thing that is bugging the heck out of me (and my users) is the "on-hold" feature. When you're on a call, and another one comes in, it doesn't ring the second line appearance on the phone, even though I have it registered separately, and I've tried to make my

We all mostly know that * as well as various SIP phones support SMS. While the final setup is somewhat of a mystery, there are reports of those lucky souls who have it working. We also know that in order to send an SMS to a mobile phone, we need to connect to some SMS message center and get the word out that way. Now, here's the new (?) element: How can I *accept* messages on my

Hi - Thanks for the fast response! > Try changing your dial statment to use a group instead such as > Dial(Zap/g2/3055551212) Same result: Spawn extension (from-sip, 000, 1) exited non-zero on 'SIP/2000-8dca' -- Executing Dial("SIP/2000-a4f1", "Zap/g1/19142246402") in new stack Dec 14 22:11:09 NOTICE[1506]: app_dial.c:800 dial_exec: Unable to create

I'm presently working on an office move and evaluation of telecommunications services needed at the new location. I'm presently wrastling with an issue related to portability and geography between landline carriers. Presently certain people within the organization are hopelessly in love with our 909-822-xxxx number(provided by pacbell/att). As that number is presently provisioned it

Hi, I have an odd problem where when a call is made to any landline phone the callerid number shows up on the caller id device with no problems. However when I call a cell phone from Sprint or Verizon it always shows up as unavaliable. I've tried multiple cell phones from each carrier. Though today I called a t-mobile phone and the callerid came through fine. Is this an * level problem

Hi - We're moving up in the world to a PRI (Verizon), and I'm having some problems with it. I'm new to this PRI thing, though, so maybe I've just screwed up a simple config detail. I've got a TE410P on a Dell PE1600SC (ServerWorks Chipset). The card itself has a green light for the PRI, and Zttool shows Span1 as "OK". All I'm getting when I call into any

I think I am missing the whole purposes of *. i see that it can do mainy things, but in laymans temrs I am not sure what it does. I am very proficient in Linux and would like to use * for the following: 1) I would like to get rid of my landline(verizon) and use voip as my main means to communicate on the telephone. I would like to be able to plug in my plain old phone into my linux box and

Hi Andres - > I have a small setup with 2 SPA3000 1 SPA2001 and 1 Polycom 301 > > The Polycom misses 1 out of 2 dialout calls, this is the full log > from a > call which didn't go through. > > 303094 Sep 14 10:45:15 DEBUG[15073]: Stopping retransmission on > 'f4e376c3-7531ff39-c86f6812@192.168.1.18' of Response 2: Found > 303095 Sep 14 10:45:15

Dear all I am going to install 2 port pri card on asterisk but i dont know how to incomming call goes in to IVR and how to route call outside base on pattern match means if some one call on mobile phone then use PRI 1 and if call on landline phon call route through pri 2 how to make dission base on pattern number Rgds satish patel

Can anybody point me to a vendor that can provide a toll free number that can be used in India to reach the united states? Verizon Business is telling me they can't get one. Thanks - Tom -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061227/f9a33cba/attachment.htm

Hello I believe that I'm getting very close in my modeling application. I've come across a challenge that I am unable to solve and would really appreciate the group's opinion. I've been using the val.prob function from the Design library (Thanks Frank!!) to both evaluate and visualize my model. From the scores and graph, it appears as my model is very accurate in

Hi, I have a client who has a system with a Sangoma 1 port PRI card with echo canceling in it. For some reason, when the system comes up the PRI will stay up for about 4-5 hours, then drop. "zap show status" shows everything as ok, but we can't make or receive any calls until the system is rebooted. Just restarting asterisk does not fix the problem. I am going to call

I'm trying to understand something that just doesn't seem to compute. How can companies like Cisco justify selling their hard phones for as much as they do? I know there is a matter of recouping R&D costs but when you look at the iPhone with all its amazing features for less than $500.00 it just doesn't make sense. Am I the only one that thinks this? Roy Anciso Director of

Hi, Has anyone on this list connected with Verizon's SIP product? We are currently undergoing interop testing with Verizon, and honestly, it seems like the most convoluted process. I'd be interested in talking with someone else who has gone through this and run a few things past you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am looking for a way to check the status of a cell phone. Found one way that worked for me and would like to have some feedback or suggestion of improvments. Below example is for a ?Swedish? cell phone, dont know if it works in the same way for other countries. I could define ?redirecting? numbers for 3 traffic cases when u dial my mobile (073-302 59 75): NOT_INUSE call forward to A INUSE

Hi John, Kevin, Tor and Wiley (and everyone else) - >> I guess the phone just doesn't register as busy when there is only one >> call on a line. It has to have two calls on a line appearance to >> register as busy. Has anyone figured out how to disable this hold >> feature and just have the second call go to the second line, the third >> call to the third line,

I have a more complicated function I am trying to write, but I run in to a problem when I want to add something to the plot from more than one data set while simultaneously controlling the appearance of the additional layer. # Toy data: foo <- data.frame ( x = 1:4, y = 4:1 , membership = c( "A", "A", "B", "B" ) ) bar <- data.frame ( x = 1:4 + 1 , y

Hi Andrew - > I have a need to be able to identify incoming calls based on some factor > (could be time of day, caller ID, dialed number, it doesn't matter.) -- > Assuming Asterisk can differentiate between the calls I want, how do I inform > the IP501? There are "only" three line appearances -- I can't simply just > ring a different appearance since there

Hi list, I'm using Asterisk2Billing (v2.0.16) and it appears to have an annoying bug. When there are rates for e.g. 44 (UK landline) and 44870 (UK premium) and a fraudster manages to somehow dial 44-870 instead of 44870 the rate for 44 will match, not the one for 44870. So, I would like to block all calls on a dialplan level that contain a dash. -44, 4-4, 44-, 44---, -, ---, just

Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4? Specifically with the Aastra 55i. Specifically, I am using the Aastra 55i with the expansion module. We want to see if other handsets are being used. (BLF) Getting BLF to work would be a great start. It sounds like setting up the hints properly will achieve this. right? Not totally sure how this should be configured.